1 files changed, 38 insertions, 87 deletions

diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index e5d120ad6..58f8a101f 100644
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -49,8 +49,6 @@ J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const FilmState> s, shared_ptr<const Op
 #ifdef HAVE_SWRESAMPLE	  
 	, _swr_context (0)
 #endif	  
-	, _deinterleave_buffer_size (8192)
-	, _deinterleave_buffer (0)
 	, _process_end (false)
 {
 	/* Create sound output files with .tmp suffixes; we will rename
@@ -68,15 +66,11 @@ J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const FilmState> s, shared_ptr<const Op
 		}
 		_sound_files.push_back (f);
 	}
-
-	/* Create buffer for deinterleaving audio */
-	_deinterleave_buffer = new uint8_t[_deinterleave_buffer_size];
 }
 
 J2KWAVEncoder::~J2KWAVEncoder ()
 {
 	terminate_worker_threads ();
-	delete[] _deinterleave_buffer;
 	close_sound_files ();
 }
 
@@ -230,14 +224,15 @@ J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio
 		stringstream s;
 		s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_sample_rate();
 		_log->log (s.str ());
-		
+
+		/* We will be using planar float data when we call the resampler */
 		_swr_context = swr_alloc_set_opts (
 			0,
 			audio_channel_layout,
-			audio_sample_format,
+			AV_SAMPLE_FMT_FLTP,
 			_fs->target_sample_rate(),
 			audio_channel_layout,
-			audio_sample_format,
+			AV_SAMPLE_FMT_FLTP,
 			_fs->audio_sample_rate(),
 			0, 0
 			);
@@ -308,14 +303,13 @@ J2KWAVEncoder::process_end ()
 #if HAVE_SWRESAMPLE	
 	if (_swr_context) {
 
+		float* out[_fs->audio_channels()];
+		for (int i = 0; i < _fs->audio_channels(); ++i) {
+			out[i] = new float[256];
+		}
+			
 		while (1) {
-			uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels()];
-			uint8_t* out[2] = {
-				buffer,
-				0
-			};
-
-			int const frames = swr_convert (_swr_context, out, 256, 0, 0);
+			int const frames = swr_convert (_swr_context, (uint8_t **) out, 256, 0, 0);
 
 			if (frames < 0) {
 				throw EncodeError ("could not run sample-rate converter");
@@ -325,7 +319,11 @@ J2KWAVEncoder::process_end ()
 				break;
 			}
 
-			write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels());
+			write_audio (out, frames);
+		}
+
+		for (int i = 0; i < _fs->audio_channels(); ++i) {
+			delete[] out[i];
 		}
 
 		swr_free (&_swr_context);
@@ -344,97 +342,50 @@ J2KWAVEncoder::process_end ()
 }
 
 void
-J2KWAVEncoder::process_audio (uint8_t* data, int size)
+J2KWAVEncoder::process_audio (float** data, int frames)
 {
-	/* This is a buffer we might use if we are sample-rate converting;
-	   it will need freeing if so.
-	*/
-	uint8_t* out_buffer = 0;
+	float* resampled[_fs->audio_channels()];
 	
-	/* Maybe sample-rate convert */
 #if HAVE_SWRESAMPLE	
+	/* Maybe sample-rate convert */
 	if (_swr_context) {
 
-		uint8_t const * in[2] = {
-			data,
-			0
-		};
+		/* Compute the resampled frames count and add 32 for luck */
+		int const resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
 
-		/* Here's samples per channel */
-		int const samples = size / _fs->bytes_per_sample();
-		
-		/* And here's frames (where 1 frame is a collection of samples, 1 for each channel,
-		   so for 5.1 a frame would be 6 samples)
-		*/
-		int const frames = samples / _fs->audio_channels();
-
-		/* Compute the resampled frame count and add 32 for luck */
-		int const out_buffer_size_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
-		int const out_buffer_size_bytes = out_buffer_size_frames * _fs->audio_channels() * _fs->bytes_per_sample();
-		out_buffer = new uint8_t[out_buffer_size_bytes];
-
-		uint8_t* out[2] = {
-			out_buffer, 
-			0
-		};
+		/* Make a buffer to put the result in */
+		for (int i = 0; i < _fs->audio_channels(); ++i) {
+			resampled[i] = new float[resampled_frames];
+		}
 
 		/* Resample audio */
-		int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, frames);
+		int out_frames = swr_convert (_swr_context, (uint8_t **) resampled, resampled_frames, (uint8_t const **) data, frames);
 		if (out_frames < 0) {
 			throw EncodeError ("could not run sample-rate converter");
 		}
 
 		/* And point our variables at the resampled audio */
-		data = out_buffer;
-		size = out_frames * _fs->audio_channels() * _fs->bytes_per_sample();
+		data = resampled;
+		frames = resampled_frames;
 	}
 #endif
 
-	write_audio (data, size);
+	write_audio (data, frames);
 
-	/* Delete the sample-rate conversion buffer, if it exists */
-	delete[] out_buffer;
+#if HAVE_SWRESAMPLE
+	if (_swr_context) {
+		for (int i = 0; i < _fs->audio_channels(); ++i) {
+			delete[] resampled[i];
+		}
+	}
+#endif	
 }
 
 void
-J2KWAVEncoder::write_audio (uint8_t* data, int size)
+J2KWAVEncoder::write_audio (float** data, int frames)
 {
-	/* XXX: we are assuming that the _deinterleave_buffer_size is a multiple
-	   of the sample size and that size is a multiple of _fs->audio_channels * sample_size.
-	*/
-
-	assert ((size % (_fs->audio_channels() * _fs->bytes_per_sample())) == 0);
-	assert ((_deinterleave_buffer_size % _fs->bytes_per_sample()) == 0);
-	
-	/* XXX: this code is very tricksy and it must be possible to make it simpler ... */
-	
-	/* Number of bytes left to read this time */
-	int remaining = size;
-	/* Our position in the output buffers, in bytes */
-	int position = 0;
-	while (remaining > 0) {
-		/* How many bytes of the deinterleaved data to do this time */
-		int this_time = min (remaining / _fs->audio_channels(), _deinterleave_buffer_size);
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			for (int j = 0; j < this_time; j += _fs->bytes_per_sample()) {
-				for (int k = 0; k < _fs->bytes_per_sample(); ++k) {
-					int const to = j + k;
-					int const from = position + (i * _fs->bytes_per_sample()) + (j * _fs->audio_channels()) + k;
-					_deinterleave_buffer[to] = data[from];
-				}
-			}
-			
-			switch (_fs->audio_sample_format()) {
-			case AV_SAMPLE_FMT_S16:
-				sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / _fs->bytes_per_sample());
-				break;
-			default:
-				throw EncodeError ("unknown audio sample format");
-			}
-		}
-		
-		position += this_time;
-		remaining -= this_time * _fs->audio_channels();
+	for (int i = 0; i < _fs->audio_channels(); ++i) {
+		sf_write_float (_sound_files[i], data[i], frames);
 	}
 }